There are two basic methods for recording sound and music—analog and digital. See e.g. Ken C. Pohlmann, “The Compact Disc: A Handbook of Theory and Use”, THE COMPUTER MUSIC AND DIGITAL AUDIO SERIES, Vol. 5 (1988). The above-mentioned audio series, which was published by A-R Editions, Inc., in Madison, Wis., is, along with all volumes therein, incorporated herein by reference.
In analog recording, the recording medium (a tape) varies continuously according to the sound signal. In other words, an analog tape stores sound signals as a continuous stream of magnetism. The magnetism, which may have any value within a limited range, varies by the same amount as the sound signal voltage.
In digital recording, the sound signal is sampled electronically and recorded as a rapid sequence of separately coded measurements. In other words, a digital recording comprises rapid measurements of a sound signal in the form of on-off binary codes represented by ones and zeros. In this digital system, zeros are represented by indentations or pits in a disc surface, and ones are represented by unpitted surfaces or land reflections of the disc, such that a compact disc contains a spiral track of binary codes in the form of sequences of minute pits produced by a laser beam.
Music that is input to a digital recording and the requisite series of reproduction processes, must pass through the recording side of a pulse code modulation (PCM) system. A master recording of the music is stored in digital form on a magnetic tape or optical disc. Once the magnetic tape has been recorded, mixed and edited, it is ready for reproduction as a CD. The CD manufacturer then converts the master tape to a master disc, which is replicated to produce a desired number of CDs. At the end of the PCM system is the reproduction side, the CD player, which outputs the pre-recorded music.
If digital technology is used in all intermediate steps between the recording and reproduction sides of the PCM system, music remains in binary code throughout the entire chain; music is converted to binary code when it enters the recording studio, and stays in binary code until it is converted back to analog form when it leaves the CD player and is audible to a listener. In most CD players, digital outputs therefrom preserve data in its original form until the data reaches the power amplifier, and the identical audio information that recorded in the studio is thereby preserved on the disc.
For example, in Prior Art FIG. 1, disclosed in U.S. Pat. No. 5,319,735, incorporated herein by reference, the four major processing components of the information embedding process that may be described as a spread spectrum technique, are shown. The process begins when digital information, comprising a sequence of code symbols to be embedded in an audio signal, is derived as output of an error control encoder 2. The error control encoder here described is generally known in the art as a Reed-Solomon encoder, which functions to improve the reliability of information retrieval.
The resulting sequence of symbols is further encoded by a spread spectrum modulator 4, which operates to produce a code signal representing a sequence of code symbols. More specifically, for instance, each code symbol produced by spread spectrum modulator 4 is represented by a pseudo-random number sequence that is filtered and modified into successive code signal segments that correspond to successive code symbols that are detectable and distinguishable using a matched filter.
Code signal shaper 6 then modifies the code signal to produce a modified code signal having frequency component levels. At each time instant, the frequency component levels are basically a pre-selected proportion of the levels of the audio signal frequency components in a corresponding frequency range. This dynamically modified code signal is subsequently combined with the original audio signal using signal combiner 8 to produce a composite audio signal. As Prior Art FIG. 1 indicates, the composite audio signal is recorded onto a recording medium, such as a digital audio tape (DAT) or, via a transmitter, is subjected to a transmission channel, which may significantly distort and/or modify the composite audio signal. This concludes the information embedding process.
Reception or playback of the composite audio signal commences the information recovery process, as illustrated in Prior Art FIG. 2, which diagrams a conventional apparatus for recovering the digital information from a composite audio signal. Upon playback from a recording medium or receiver, the composite audio signal is transformed into an equalized signal by signal equalizer 10. The code signal detector and synchronizer 12 immediately detects the presence of the code signal in the newly transformed equalized signal, just as the spread spectrum demodulator 14 recovers the sequence of code symbols from the equalized signal. Consequently, an error control decoder 16, such as a Reed-Solomon decoder 16, recovers the signaled digital information from the sequence of code symbols.
Examining the spread spectrum modulator 4 in more detail, see Prior Art FIG. 3, each code symbol is input into a pseudo-random number sequence generator 18 where each possible input code symbol uniquely corresponds to a designated pseudo-random number sequence. By way of illustration, the collection of pseudo-random number sequences is stored in a lookup table so that each successive input code symbol selects a corresponding pseudo-random number sequence, which is produced by generator 18 by retrieving this pseudo-random number sequence from the lookup table. Alternatively, pseudo-random number sequence generator 18 may be designed as a special purpose circuit to dynamically produce the pseudo-random number sequences that correspond to successive input code symbols.
The pseudo-random number sequences fed to the unsampler 20 as a digital signal are then unsampled to a higher clock frequency. The resulting higher rate digital signal is low-pass filtered by low pass filter 22 in order to eliminate extraneous high frequency components. This process generates a baseband code signal that is frequency shifted to a signaling band by the frequency shift-to-signaling band 24. The signaling band, which ranges from 1890 Hz to 10710 Hz, lies within the bandwidth of the audio signal.
Another major component of the information embedding process, the code signal shaper 6, is now described in more detail in Prior Art FIG. 4. Here, the original audio signal (i.e. the music), into which the code signal is to be embedded, is manipulated by frequency analyzer 26, and is continuously frequency analyzed over a frequency band encompassing the signaling band. The energy calculation procedure 28 uses the result of this analysis to calculate a frequency distribution of the audio signal masking energy as it evolves over time. This frequency distribution of code signal energy is calculated by the relative masking level gain calculation procedure 30, which produces a set of gain values. Each gain value corresponds to a distinct frequency range within the signaling band.
At this juncture, the signal shaper 32 selectively filters and decomposes the code signal into component signals that occupy distinct frequency ranges. This is accomplished by using the gain values calculated by procedure 30 to adjust the levels of the corresponding component signals; these adjusted signals are then combined to produce the modified code signal. “The overall effect of this procedure is to produce a dynamically modified code signal with frequency component levels which are, at each time instant, essentially a preselected proportion of the levels of the audio signal frequency components in the corresponding frequency range.” See U.S. Pat. No. 5,319,735, col. 6, lines 49–54.
Yet another major component of the information embedding process is the signal combiner 8, which is described in more detail in Prior Art FIG. 5. The signal combiner 8 comprises a signal delay 34 and a signal adder 36. The primary function of the signal delay 34 is to compensate for delays produced by the code signal shaper 6. Consequently, signal delay 34 delays the original audio signal. “This compensating delay temporally aligns the original audio signal with the modified code signal . . . when these two signals are added or combined in block [36] to form the composite audio signal.” Id. at lines 63–68.
The frequency components of this alignment are illustrated in Prior Art FIG. 6. Here it is seen that the frequency distribution of the modified code signal energy basically parallels the frequency distribution of the original audio signal energy by a fixed offset, within the signaling band. The offset, which is measured in decibel (dB) units, is generally referred to as the Code to Music Ratio (CMR). The CMR is a preselected value that determines the ability to distinguish a composite audio signal from the original audio signal by listening. A conservative nominal design value for the CMR that renders a composite audio signal virtually indistinguishable from the original audio signal by listening, is−19 dB.
The fourth and final major component of the information embedding process is the error control encoder 2, as illustrated in more detail in Prior Art FIG. 7. A digital music source 38, which may be represented by an optical disc or a continuing stream of music information bits, is operated along with coded message 40, which represents the code signal to be embedded. The processing that occurs within the encoder, as at block 42, includes generating a continuing message sequence of 1s and 0s. The transmitted sequence may or may not be longer than the message sequence.
The processed message 40, generally composed of 16 bits and already frequency analyzed, enters an adjust gain device 44, which functions to adjust the original music signal and message 40 by employing an adding ratio of gain values. The resulting adjusted signal introduced into the signal combiner device 46 is output as a modified code signal 48 comprising the original music signal and the embedded message; this output 48 comprises 16 bits also.
Thus, generally when digital information, like music, is to be signaled for an embedding process, the digital information is transformed, using a conventional spread spectrum technique, into a modified code signal. Through the various processing steps of the spread spectrum technique, this code signal is modified in such a way that the modified code signal can be combined with the original digitized music information to form a composite audio signal that is indistinguishable from the original audio signal by normal listening.
At this point, the modified code signal can be recorded, or subjected to a transmission channel, that generally distorts and/or modifies the composite audio signal. The digital information, which is represented by a sequence of code symbols that is filtered and dynamically modified to be detectable by a matched filter, can then be recovered from the distorted and/or modified composite audio signal.
The following additional prior art patent represents the general state of embedded signaling, and is hereby incorporated by reference:
U.S. Pat. No. 4,914,439 to Nakahashi et al. discloses an analog to digital conversion system using dithering technology.
One problem basic to the above prior art references involves low level signals. For instance, when music or any audio signal, into which data is to be embedded, begins the information embedding process at a very low audio level, data and processing are generally added to the music signal at approximately−19 to−20 dB, which is already 20 decibels below music level. Thus, when the music level goes down to nearly zero, which can happen with high quality music on CDs, the problem of ineffective embedding is exacerbated; proper receipt of the music, as well as addition of the data to the music signal, is jeopardized.
Accordingly, I have determined that the above prior art methods are ineffective when the data being embedded together includes low level audio data. In addition, the above prior art references do not provide an inexpensive, easily adaptable and/or compatible system or method of embedding signals, particularly, for example, low level audio signals.